Freepbx outboud/inbound calls not working with freephoneline.ca

General Help
1

Hello

I have recently installed the FreePBX on my Raspberry PI. I am using trunks from freephoneline.ca. I have sip clients on my android phone and soft client on my pc.

I know my trunks are configured and connected to freephoneline.ca from my FreepbxRPI. On the freephoneline.ca, I see the status is connected.

I am using a ddns service to register the pbx ip address.

My sip clients can call each other no problems locally using extentions. I know they are configured correctly to connect to freePBX. My android device with CSipSimple, the client will register to my FreePBX when on WIFI and on 3G.

My problem is…

I am still unable to receive inbound calls or place outbound calls.

What am I missing.

I am not sure what I need to post here to give more information. I can ssh into the Freepbx and get more info as required.

2

Asterisk 11.6.0 built by root @ raspbx on a armv6l running Linux on 2013-11-05 22:19:11 UTC

3
  • Name : freephoneline
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-trunk-sip-freephoneline
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : No
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Force rport : Yes
    Symmetric RTP: Yes
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : Yes
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : voip.freephoneline.ca
    Addr->IP : 208.65.240.44:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: 1XXXXXX8237
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw|g729)
    Codec Order : (ulaw:20,g729:20,gsm:20,alaw:20)
    Auto-Framing : No
    Status : Unmonitored
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
4
  • Name : freephoneline
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-trunk-sip-freephoneline
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : No
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Force rport : Yes
    Symmetric RTP: Yes
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : Yes
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : voip.freephoneline.ca
    Addr->IP : 208.65.240.44:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: 1XXXXXX8237
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw|g729)
    Codec Order : (ulaw:20,g729:20,gsm:20,alaw:20)
    Auto-Framing : No
    Status : Unmonitored
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
5

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.11.0(11.6.0)
SDP Session Name: Asterisk PBX 11.6.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externhost
Externhost: ######.#####ddns.com
Externaddr: XXX.19.191.41:0
Externrefresh: 120
Localnet: 192.168.0.0/255.255.255.0

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|g729)
Codec Order: ulaw:20,g729:20,gsm:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 180 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 180 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 180 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

6

[2014-02-17 00:03:58] VERBOSE[6906][C-00000002] netsock2.c: == Using SIP RTP TOS bits 184
[2014-02-17 00:03:58] VERBOSE[6906][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5
[2014-02-17 00:03:58] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:1] ResetCDR(“SIP/228-00000001”, “”) in new stack
[2014-02-17 00:03:58] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:2] NoCDR(“SIP/228-00000001”, “”) in new stack
[2014-02-17 00:03:58] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:3] Progress(“SIP/228-00000001”, “”) in new stack
[2014-02-17 00:03:58] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:4] Wait(“SIP/228-00000001”, “1”) in new stack
[2014-02-17 00:03:59] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:5] Progress(“SIP/228-00000001”, “”) in new stack
[2014-02-17 00:03:59] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:6] Playback(“SIP/228-00000001”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2014-02-17 00:03:59] VERBOSE[6959][C-00000002] file.c: – <SIP/228-00000001> Playing ‘silence/1.ulaw’ (language ‘en’)
[2014-02-17 00:04:00] VERBOSE[6959][C-00000002] file.c: – <SIP/228-00000001> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2014-02-17 00:04:02] VERBOSE[6959][C-00000002] file.c: – <SIP/228-00000001> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2014-02-17 00:04:05] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:7] Wait(“SIP/228-00000001”, “1”) in new stack
[2014-02-17 00:04:06] VERBOSE[6959][C-00000002] pbx.c: – Executing [6046826644@from-internal:8] Congestion(“SIP/228-00000001”, “20”) in new stack
[2014-02-17 00:04:06] VERBOSE[6959][C-00000002] pbx.c: == Spawn extension (from-internal, 6046826644, 8) exited non-zero on ‘SIP/228-00000001’
[2014-02-17 00:04:06] VERBOSE[6959][C-00000002] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/228-00000001”, “”) in new stack
[2014-02-17 00:04:06] VERBOSE[6959][C-00000002] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/228-00000001’
[2014-02-17 00:04:09] ERROR[6960] tcptls.c: Unable to connect SIP socket to 199.7.159.41:58277: Connection refused

7

Bump

8

I don’t think your issue is trunk/itsp related…I don’t even see a dial attempt in your log above. Are you sure the basics of your dialplan (trunks/routes/extensions) are setup right?

If so, repost a log entry from the point of dial until hangup and I’ll take a closer look. But I don’t think you’re looking in the right place.